IP Packet charging for multimedia services

II.4.3 BASIC QoS ARCHITECTURE

The basic architecture introduces the three fundamental pieces for QoS implementation

· QoS identification and marking techniques for coordinating QoS from end to end between network elements

· QoS within a single network element (for example, queuing, scheduling, and trafficshaping tools)

· QoS policy, management, and accounting functions to control and administer end-to-end traffic across a network

Figure 11: A Basic QoS Implementation Has Three Main Components

Source: http://www.cisco.com/univercd/cc/td/doc/cisintwk/ito_doc/qos.htm,september 24,2006

II.4.4 QoS WITHIN A SINGLE NETWORK ELEMENT

Congestion management, queue management, link efficiency, and shaping/policing tools provide QoS within a single network element.

II.4.4.1 Congestion Management

Because of the bursty nature of voice/video/data traffic, sometimes the amount of traffic exceeds the speed of a link. At this point, some question has been asked, such as:

· What will the router do?

· Will it buffer traffic in a single queue and let the first packet in be the first packet out?

· Or, will it put packets into different queues and service certain queues more often?

Congestion-management tools address these questions. Tools include priority queuing (PQ), custom queuing (CQ), weighted fair queuing (WFQ), and class-based weighted fair queuing (CBWFQ).

II.4.4.2 Queue Management

Because queues are not of infinite size, they can fill and overflow. When a queue is full, any additional packets cannot get into the queue and will be dropped. This is a tail drop. The issue with tail drops is that the router cannot prevent this packet from being dropped (even if it is a high-priority packet). So, a mechanism is necessary to do two things:

1. Try to make sure that the queue does not fill up, so that there is room for high-priority packets

2. Allow some sort of criteria for dropping packets that are of lower priority before dropping higher-priority packets Weighted early random detect (WRED) provides both of these mechanisms.¹⁶

16 http://www.cisco.com/univercd/cc/td/doc/cisintwk/ito_doc/qos.htm, September 24, 2006

II.5 MULTIMEDIA OVER IP

In this point let talk more about the following protocols:

> Resource ReServVation Protocol (RSVP) > Real-Time Transport Protocol (RTP)

> Real-Time Control Protocol (RTCP)

> Real-Time Streaming Protocol (RTSP)

Which are the foundations ofreal-time services. II.5.3.1 RSVP (Resource ReSerVation Protocol)

RSVP is used to set up reservations for network resources. When an application in a host (the data stream receiver) requests a specific quality of service (QoS) for its data stream, it uses RSVP to deliver its request to routers along the data stream paths. RSVP is responsible for the negotiation of connection parameters with these routers. If the reservation is setup, RSVP is also responsible for maintaining router and host states to provide the requested service. Each node capable of resource reservation has several local procedures for reservation setup and enforcement

II.5.3.2 RTP (Real-time Transport Protocol)

Realtime transport protocol (RTP) is an IP-based protocol providing support for the transport of real-time data such as video and audio streams. The services provided by RTP include time reconstruction, loss detection, security and content identification.

RTP is primarily designed for multicast of real-time data, but it can be also used in unicast. It can be used for one-way transport such as video-on-demand as well as interactive services such as Internet telephony.

II.5.3.3 RTCP (Real-Time Control Protocol)

RTCP is the control protocol designed to work in conjunction with RTP.In an RTP session, participants periodically send RTCP packets to convey feedback on quality of data delivery and information of membership. There are five RTCP packet types to carry control information. These five types are:

> Receiver Report (RRe): Receiver reports are generated by participants that are not active senders. They contain reception quality feedback about data delivery, including the highest packets number received, the number of packets lost, inter-arrival jitter, and timestamps to calculate the round-trip delay between the sender and the receiver.

> Sender Report (SR): Sender reports are generated by active senders. In addition to the reception quality feedback as in RR, they contain a sender information section, providing information on inter-media synchronization, cumulative packet counters, and number of bytes sent.

> Source description items (SDES): They contain information to describe the sources. > BYE: indicates end of participation.

> APP: application specific functions. It is now intended for experimental use as new applications and new features are developed.

II.5.3.4 Real-Time Streaming Protocol (RTSP)

Instead of storing large multimedia files and playing back, multimedia data is usually sent across the network in streams. Streaming breaks data into packets with size suitable for transmission between the servers and clients.

The real-time data flows through the transmission, decompressing and playing back pipeline just like a water stream. A client can play the first packet; decompress the second, while receiving the third. Thus the user can start enjoying the multimedia without waiting to the end of transmission.

RTSP, the Real Time Streaming Protocol, is a client-server multimedia presentation protocol to enable controlled delivery of streamed multimedia data over IP network. It provides "VCR-style" remote control functionality for audio and video streams, like pause, fast forward, reverse, and absolute positioning. Sources of data include both live data feeds and stored clips.

RTSP is an application-level protocol designed to work with lower-level protocols like RTP, RSVP to provide a complete streaming service over internet. It provides means for choosing delivery channels (such as UDP, multicast UDP and TCP), and delivery mechanisms based upon RTP. It works for large audience multicast as well as single-viewer unicast.

> MULTIMEDIA NETWORKING

Multimedia can roughly be defined as a technology that enables humans to use computers capable of processing textual data, audio and video, still pictures, and animation. Applications range over entertainment, education, information provision; design e.g. CAD/CAM, co-operative working such as video conferencing, application sharing, remote working and virtual reality experiences.

Multimedia applications for computers have been developed for single computing platforms such as the PC, Apple Mac and games machines. The importance of communications or networking for multimedia lies in the new applications that will be generated by adding networking capabilities to multimedia computers, and hopefully gains in efficiency and cost of ownership and use when multimedia resources are part of distributed computing systems. Widening of access to multimedia sources and potential markets in multimedia, video and information are commercial driving force for networking multimedia.

The reality of networking multimedia is that, the characteristics of multimedia make heavy demands on storage and transmission systems. Data compression can be used to reduce the demands of multimedia, particularly of video and audio on these systems, but usually at the expense of some loss in the detail compared with the source and at extra cost.

The ways in which users or participants in multimedia sessions access multimedia or connect with others have important consequences for the storage and transmission systems. For instance multimedia learning material can be accessed directly from a server during a class or downloaded to student machines prior to a session. The demands on a connecting network are very different in each access mode.

The cost of transmitting multimedia information will determine the pace of development of networked multimedia applications. The availability of standards for multimedia networking, particularly for inter-working between applications, the development of networked applications, and interworking between networks are essential to reduce the complexity and level of skill required in using multimedia.

II.5.1.1 USER REQUIREMENTS FOR MULTIMEDIA II.5.1.1.1 Computer Interface

The standards of reproduction for computers which are desirable have been set by the publishers of books, music, Walt Disney cartoons and television producers. With the development of High Definition TV and beyond, it is likely that there will be a continual increase in the demands placed on computer based multimedia systems.

The current PAL standard in the UK delivers video in 625 lines at 25 frames/sec. High Definition TV delivers video in 1250 lines with a higher horizontal resolution at 25 frames/sec and requires about five times the information rate as the current PAL system.

Multimedia applications like any other application, appliance or tool, benefit from being easy to use, with minimal training or self learning. The need for a well designed human - computer interface, which may be screen or audio based is well accepted.

II.5.1.1.2 Access, Delivery, Scheduling and Recording

Alternatively the delivery of information at a later time is acceptable if it can be scheduled, as in a TV broadcast schedule, or a first class postal letter. Scheduling the delivery of multimedia information has not been widely implemented. Scheduling can have advantages for users over on demand delivery. In a learning situation times can be defined for class attendance by a lecturer. In open learning situations learners can control their programme by requesting a multimedia unit at a convenient time.

Just as we can record a TV film on a VHS recorder, some multimedia computer users will wish to record a film, session, or learning experience for future reference.